Voice-Over-Internet Protocol (VoIP) is attracting a multitude of users because VoIP offers tremendous cost savings relative to a Public Switching Telephone Network (PSTN). For instance, users may bypass long-distance carriers with per minute charges in lieu of transmitting voice calls over the Internet for a flat monthly Internet access fee.
Internet telephony within an intranet enables users to reduce costs by eliminating long-distance charges between sites included in the intranet. An intranet is a local-area network which may or may not be connected to the Internet, but which has some similar functions. The intranet is used for connectivity within, for example, a company. Some companies set up World Wide Web servers on their own internal networks so employees have access to the organization's Web documents. Users may make point-to-point calls via gateway servers attached to a local-area network. For example, a user may want to make a point-to-point call to another user in another office included in the same intranet. The calling party will dial an extension to connect with the gateway server, which is equipped with a telephony board and compression-conversion software; the server configures a private branch exchange (PBX) to digitize the upcoming call. The calling party then dials the number of the called party and the gateway server transmits the call over the IP-based wide-area network to the gateway to the destination office. The destination gateway converts the digital signal back to analog format and delivers the call to the called party.
Although progressing rapidly, VoIP continues to exhibit decreased reliability and sound quality when compared to the PSTN, due primarily to limitations both in Internet bandwidth, current compression technology, delay, jitter, and packet loss. Because the Internet is a packet-switched network, the individual packets of each voice signal may travel over separate network paths for reassembly in the proper sequence at the destination. Although transmitting each packet over a separate path creates a high efficiency for network resources over the PSTN, the chances for packet loss also increase. Packet loss shows up in the form of gaps or periods of silence in a conversation, leading to a clipped-speech effect that is unsatisfactory for most users and unacceptable in business communications. As a result, most corporations looking to reduce communication costs confine their Internet-telephony applications to their intranets. With more predictable bandwidth available than the public Internet, intranets can support full-duplex, real-time voice communications. However, restricting Internet telephony to company intranets does not allow optimum cost saving benefits or flexibility when compared to Internet-telephony over the public Internet.
To date, most developers of Internet-telephony software, as well as vendors of gateway servers, have been using a variety of speech-compression protocols. The use of various speech-coding algorithms, with different bit rates and mechanisms for reconstructing voice packets and handling delays, produces varying levels of intelligibility and fidelity in sound transmitted over the public Internet.
An evolving solution to the varying levels of quality of sound, etc. transmitted over the Internet is to tier the public Internet. Users of the public Internet will then be required to pay for the specific service levels or Quality of Service (QoS) they require. A Service Level Agreement (SLA) is a contract between a carrier and a customer that defines the terms of the carrier's responsibility to the customer and the type and extent of remuneration if those responsibilities are not met. Reports on the QoS based on either per-call measurements or per-path measurements are a tool for determining if carrier responsibilities are met and if not, any rebate due to the customer. Per-call measurements are capable of illustrating voice quality on a call-by-call basis, which more closely reflects the customer's calling experience. However, in many cases, VoIP gateways or IP-PBXs are managed by the customer and therefore per-call information is not available. For instance, when a customer manages the VoIP gateways, the customer security restrictions or technical constraints may prevent the dissemination of per-call information.
Currently, one solution requires the installation of additional hardware at each customer site to take performance measurements. This solution is not scalable and due to the excessive hardware costs, additional cost of maintaining equipment, and additional network connectivity required to communicate and support the additional hardware, most cost-conscious users would not implement the additional hardware.
Therefore, there is a need for a system and method for making path-based VoIP quality measurements without deploying additional hardware at each customer site.